October 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: wm_adsp: Remove duplicate set of kcontrol->iface" to the asoc tree
by Mark Brown 05 Oct '16

05 Oct '16

The patch ASoC: wm_adsp: Remove duplicate set of kcontrol->iface has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e23cd27da59cd2245fc4450f183e4c1223021eca Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Date: Mon, 3 Oct 2016 11:40:38 +0100 Subject: [PATCH] ASoC: wm_adsp: Remove duplicate set of kcontrol->iface Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b943dde8dbe5..7320fcacd86a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -967,7 +967,6 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); if (!kcontrol) return -ENOMEM; - kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->name = ctl->name; kcontrol->info = wm_coeff_info; -- 2.9.3

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[alsa-devel] [PATCH] ASoC: sun4i-i2s: fix odd_ptr_err.cocci warnings
by Julia Lawall 05 Oct '16

05 Oct '16

sound/soc/sunxi/sun4i-i2s.c:633:5-11: inconsistent IS_ERR and PTR_ERR on line 635. PTR_ERR should normally access the value just tested by IS_ERR Generated by: scripts/coccinelle/tests/odd_ptr_err.cocci Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- Please take the patch only if it's a positive warning. Thanks! sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -632,7 +632,7 @@ static int sun4i_i2s_probe(struct platfo reset_apb = devm_reset_control_get(&pdev->dev, "apb_reset"); if (IS_ERR(reset_apb)) { dev_err(&pdev->dev, "Can't get apb reset\n"); - return PTR_ERR(i2s->mod_clk); + return PTR_ERR(reset_apb); } ret = reset_control_deassert(reset_apb);

2 1

[alsa-devel] [PATCH 0/6] Arizona FLL fixes and Clocking Improvements
by Charles Keepax 04 Oct '16

04 Oct '16

Hi, The first 3 patches in this series fix some small issues with the FLL configuration in the driver. The second 3 patches add support for requesting the input clocks for the CODEC through the clock framework. This is some what of an intermediate step until all the clocking itself in the CODEC can be brought under the clock framework but as there are machine drivers that want to be merged now with source clocks that require gating and difficult to resolve issues between the SPI and clock framework an intermediate solution seems warranted. Whilst the two halfs of this series do fairly different things, I have sent them as once series as they touch a lot of the same code, so this seemed the simpler approach to pushing them both through. The series is based of Mark's ASoC for-next tree. Thanks, Charles Charles Keepax (5): ASoC: arizona: Correct handling of FLL theta in synchroniser mode ASoC: arizona: Allow specification of base for arizona_is_enabled_fll ASoC: arizona: Avoid changing SYNC_ENA whilst the FLL_ENA is set ASoC: arizona: Add gating for clock when used for direct MCLK ASoC: arizona: Add gating for source clocks of the FLLs Sylwester Nawrocki (1): mfd: arizona: Add gating of external MCLKn clocks drivers/mfd/arizona-core.c | 30 +++++++++- include/linux/mfd/arizona/core.h | 9 +++ sound/soc/codecs/arizona.c | 124 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs47l24.c | 6 +- sound/soc/codecs/wm5102.c | 9 ++- sound/soc/codecs/wm5110.c | 10 +++- sound/soc/codecs/wm8997.c | 9 ++- sound/soc/codecs/wm8998.c | 6 +- 9 files changed, 179 insertions(+), 26 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: wm9712: Convert to regmap
by Lars-Peter Clausen 04 Oct '16

04 Oct '16

Currently the wm9712 driver still uses custom snd_soc_codec_driver IO callbacks. This has been deprecated for a while, so convert the wm9712 driver to use regmap for its IO. Tested-by: Marek Vasut <marex(a)denx.de> Acked-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9712.c | 143 +++++++++++++++++++++++++--------------------- 2 files changed, 79 insertions(+), 65 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c67667b..f472254 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1057,6 +1057,7 @@ config SND_SOC_WM9705 config SND_SOC_WM9712 tristate + select REGMAP_AC97 config SND_SOC_WM9713 tristate diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 557709e..d2d0d2b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -32,31 +33,66 @@ struct wm9712_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); +static const struct reg_default wm9712_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x08, 0x0f0f }, + { 0x0a, 0xaaa0 }, + { 0x0c, 0xc008 }, + { 0x0e, 0x6808 }, + { 0x10, 0xe808 }, + { 0x12, 0xaaa0 }, + { 0x14, 0xad00 }, + { 0x16, 0x8000 }, + { 0x18, 0xe808 }, + { 0x1a, 0x3000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x22, 0x0000 }, + { 0x26, 0x000f }, + { 0x28, 0x0605 }, + { 0x2a, 0x0410 }, + { 0x2c, 0xbb80 }, + { 0x2e, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x34, 0x2000 }, + { 0x4c, 0xf83e }, + { 0x4e, 0xffff }, + { 0x50, 0x0000 }, + { 0x52, 0x0000 }, + { 0x56, 0xf83e }, + { 0x58, 0x0008 }, + { 0x5c, 0x0000 }, + { 0x60, 0xb032 }, + { 0x62, 0x3e00 }, + { 0x64, 0x0000 }, + { 0x76, 0x0006 }, + { 0x78, 0x0001 }, + { 0x7a, 0x0000 }, +}; -/* - * WM9712 register cache - */ -static const u16 wm9712_reg[] = { - 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */ - 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */ - 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */ - 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */ - 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ - 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */ - 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ - 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */ - 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */ - 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */ - 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */ - 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ - 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ - 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ +static bool wm9712_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AC97_REC_GAIN: + return true; + default: + return regmap_ac97_default_volatile(dev, reg); + } +} + +static const struct regmap_config wm9712_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm9712_volatile_reg, + + .reg_defaults = wm9712_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9712_reg_defaults), }; #define HPL_MIXER 0x0 @@ -488,35 +524,13 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_REC_GAIN) - return soc_ac97_ops->read(wm9712->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9712_reg))) - return -EIO; - - return cache[reg]; - } + return snd_soc_read(codec, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - soc_ac97_ops->write(wm9712->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9712_reg))) - cache[reg] = val; - - return 0; + return snd_soc_write(codec, reg, val); } static int ac97_prepare(struct snd_pcm_substream *substream, @@ -619,8 +633,7 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_soc_resume(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); @@ -629,15 +642,8 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret == 0) { - /* Sync reg_cache with the hardware after cold reset */ - for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) { - if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || - (i > 0x58 && i != 0x5c)) - continue; - soc_ac97_ops->write(wm9712->ac97, i, cache[i>>1]); - } - } + if (ret == 0) + regcache_sync(codec->component.regmap); return ret; } @@ -645,6 +651,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) static int wm9712_soc_probe(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap; int ret; wm9712->ac97 = snd_soc_new_ac97_codec(codec, WM9712_VENDOR_ID, @@ -655,16 +662,28 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) return ret; } + regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97_codec; + } + + snd_soc_codec_init_regmap(codec, regmap); + /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; +err_free_ac97_codec: + snd_soc_free_ac97_codec(wm9712->ac97); + return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9712->ac97); return 0; } @@ -673,14 +692,8 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .probe = wm9712_soc_probe, .remove = wm9712_soc_remove, .resume = wm9712_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9712_set_bias_level, .suspend_bias_off = true, - .reg_cache_size = ARRAY_SIZE(wm9712_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9712_reg, .component_driver = { .controls = wm9712_snd_ac97_controls, -- 2.1.4

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[alsa-devel] [PATCH] ASoC: wm_adsp: Remove duplicate set of kcontrol->iface
by Richard Fitzgerald 03 Oct '16

03 Oct '16

Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b943dde..7320fca 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -962,17 +962,16 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) int ret; if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); if (!kcontrol) return -ENOMEM; - kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->name = ctl->name; kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; kcontrol->tlv.c = snd_soc_bytes_tlv_callback; kcontrol->private_value = (unsigned long)&ctl->bytes_ext; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: da7219: Improve pop/click performance for sensitive HPs
by Adam Thomson 03 Oct '16

03 Oct '16

Currently on some headsets slight pops can be heard during DAPM power-up/down. This can also be witnessed during the HP detect procedure. This patch addresses the issue by adjusting DAPM power sequencing slightly, the introduction of delays and use of minimum HP gain to avoid such noise artefacts. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes are based on top of commit: ASoC: da7219: Disable AAD if codec is not a wake-up source sound/soc/codecs/da7219-aad.c | 18 ++++-- sound/soc/codecs/da7219.c | 139 ++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/da7219.h | 5 ++ 3 files changed, 140 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 2b8914d..6274d79 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -204,10 +204,19 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, DA7219_MIXOUT_R_AMP_EN_MASK, DA7219_MIXOUT_R_AMP_EN_MASK); - snd_soc_write(codec, DA7219_HP_L_CTRL, - DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); - snd_soc_write(codec, DA7219_HP_R_CTRL, - DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + msleep(DA7219_SETTLING_DELAY); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_MUTE_EN_MASK | + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_MUTE_EN_MASK | + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, 0); /* * If we're running from the internal oscillator then give audio paths @@ -244,6 +253,7 @@ static void da7219_aad_hptest_work(struct work_struct *work) regcache_mark_dirty(da7219->regmap); regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, DA7219_HP_R_CTRL); + msleep(DA7219_SETTLING_DELAY); regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, DA7219_MIXOUT_R_CTRL); regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 65f7e98..7c26ee3 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -823,6 +823,85 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, } } +static int da7219_settling_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + msleep(DA7219_SETTLING_DELAY); + break; + default: + break; + } + + return 0; +} + +static int da7219_mixout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + u8 hp_ctrl, min_gain_mask; + + switch (w->reg) { + case DA7219_MIXOUT_L_CTRL: + hp_ctrl = DA7219_HP_L_CTRL; + min_gain_mask = DA7219_HP_L_AMP_MIN_GAIN_EN_MASK; + break; + case DA7219_MIXOUT_R_CTRL: + hp_ctrl = DA7219_HP_R_CTRL; + min_gain_mask = DA7219_HP_R_AMP_MIN_GAIN_EN_MASK; + break; + default: + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + /* Enable minimum gain on HP to avoid pops */ + snd_soc_update_bits(codec, hp_ctrl, min_gain_mask, + min_gain_mask); + + msleep(DA7219_MIN_GAIN_DELAY); + + break; + case SND_SOC_DAPM_POST_PMU: + /* Remove minimum gain on HP */ + snd_soc_update_bits(codec, hp_ctrl, min_gain_mask, 0); + + break; + } + + return 0; +} + +static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_PRE_PMD: + /* Ensure nominal gain ramping for DAPM sequence */ + da7219->gain_ramp_ctrl = + snd_soc_read(codec, DA7219_GAIN_RAMP_CTRL); + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, + DA7219_GAIN_RAMP_RATE_NOMINAL); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Restore previous gain ramp settings */ + snd_soc_write(codec, DA7219_GAIN_RAMP_CTRL, + da7219->gain_ramp_ctrl); + break; + } + + return 0; +} + /* * DAPM Widgets @@ -906,30 +985,46 @@ static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), /* DACs */ - SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, - DA7219_NO_INVERT), - SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, - DA7219_NO_INVERT), + SND_SOC_DAPM_DAC_E("DACL", NULL, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DACR", NULL, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output PGAs */ - SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, - DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, - DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, - DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), - SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, - DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA_E("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mixout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mixout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY_S("Headphone Left PGA", 1, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("Headphone Right PGA", 1, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Output Supplies */ - SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, - DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 0, DA7219_CP_CTRL, + DA7219_CP_EN_SHIFT, DA7219_NO_INVERT, + da7219_settling_event, + SND_SOC_DAPM_POST_PMU), /* Outputs */ SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), + + /* Pre/Post Power */ + SND_SOC_DAPM_PRE("Pre Power Gain Ramp", da7219_gain_ramp_event), + SND_SOC_DAPM_POST("Post Power Gain Ramp", da7219_gain_ramp_event), }; @@ -1002,8 +1097,8 @@ static const struct snd_soc_dapm_route da7219_audio_map[] = { {"Mixout Left PGA", NULL, "DACL"}, {"Mixout Right PGA", NULL, "DACR"}, - {"Headphone Left PGA", NULL, "Mixout Left PGA"}, - {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + {"HPL", NULL, "Mixout Left PGA"}, + {"HPR", NULL, "Mixout Right PGA"}, {"HPL", NULL, "Headphone Left PGA"}, {"HPR", NULL, "Headphone Right PGA"}, @@ -1711,6 +1806,14 @@ static int da7219_probe(struct snd_soc_codec *codec) DA7219_HP_R_AMP_RAMP_EN_MASK, DA7219_HP_R_AMP_RAMP_EN_MASK); + /* Default minimum gain on HP to avoid pops during DAPM sequencing */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK, + DA7219_HP_L_AMP_MIN_GAIN_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK, + DA7219_HP_R_AMP_MIN_GAIN_EN_MASK); + /* Default infinite tone gen, start/stop by Kcontrol */ snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 66d3bad..6baba74 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -777,6 +777,10 @@ #define DA7219_SYS_STAT_CHECK_RETRIES 6 #define DA7219_SYS_STAT_CHECK_DELAY 50 +/* Power up/down Delays */ +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 + enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, DA7219_CLKSRC_MCLK_SQR, @@ -814,6 +818,7 @@ struct da7219_priv { bool master; bool alc_en; + u8 gain_ramp_ctrl; }; #endif /* __DA7219_H */ -- 1.9.3

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[alsa-devel] [RFC PATCH v2 0/2] support DP MST audio
by libin.yang＠linux.intel.com 01 Oct '16

01 Oct '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-intel-nightly on git://anongit.freedesktop.org/drm-intel Libin Yang (2): ALSA: hda - add DP mst verb support ALSA: hda - add DP mst audio support sound/pci/hda/hda_codec.c | 77 +++++++++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 216 ++++++++++++++++++++++++++++++++++----------- 3 files changed, 241 insertions(+), 55 deletions(-) -- 1.9.1

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